Packet-based data networks are widely used to link various types of network elements, such as personal computers, servers, gateways, network telephones, and so forth. Data networks may include private networks (such as local area networks or wide area networks) and public networks (such as the Internet). Popular forms of communications between network elements across packet-based data networks include electronic mail, file transfer, web browsing, and other exchanges of digital data.
With the increased capacity and reliability of packet-based data networks, voice communications (including telephone calls, video conferencing, and so forth) over data networks have become possible. Voice communications over data networks are unlike voice communications in a conventional public-switched telephone network (PSTN), which provides users with dedicated, end-to-end circuit connections for the duration of each call. Communications over data networks, such as IP (Internet Protocol) networks, are performed using packets or datagrams that are sent in bursts from a source to one or more destination nodes. Voice data sent over a data network typically shares network bandwidth with conventional non-voice data (e.g., data associated with electronic mail, file transfer, web access, and other traffic).
Various standards have been proposed for voice and multimedia communications over data networks. One such standard is the H.323 Recommendation from the International Telecommunications Union (ITU), which describes terminals, equipment, and services for multimedia communications over data networks.
Another standard for voice and multimedia communications is the Session Initiation Protocol (SIP), which establishes, maintains, and terminates multimedia sessions over a data network. SIP is part of a multimedia data and control architecture developed by the Internet Engineering Task Force (IETF). The IETF multimedia data and control architecture also includes other protocols to enable voice and multimedia sessions over data networks, including the Resource Reservation Protocol (RSVP) for reserving network resources; the Real-Time Transport Protocol (RTP) for transporting real-time data and providing quality of service (QoS) feedback; the Real-Time Streaming Protocol (RTSP) for controlling delivery of streaming media; the Session Announcement Protocol (SAP) for advertising multimedia sessions by multicast; and the Session Description Protocol (SDP) for describing multimedia sessions.
In the burgeoning field of Internet telephony, the ability to provide calling party information to a receiving party while establishing a call session is somewhat limited. The calling party information may, in one embodiment, include any information that may be somewhat associated with a party that originates a call request. Access to such information, in some instances, may be useful, and perhaps desirable, to a party receiving the call request.
A need thus exists for providing calling party information to a receiving party in a request to establish a call session.